Chromium Code Reviews| Index: webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc |
| diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc |
| index 855a39e6752bd5fd4caa7b191aa2a783eedf9f60..ec3f4fcddc22dc7a5466f68d71e1d8f9629b0dfb 100644 |
| --- a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc |
| +++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc |
| @@ -13,6 +13,8 @@ |
| #include <assert.h> |
| #include <stdio.h> |
| +#include <memory> |
|
hlundin-webrtc
2016/02/15 11:18:15
You do not have to repeat the same include in both
kwiberg-webrtc
2016/02/15 11:42:50
Yeah, the script I whipped up for automating this
|
| + |
| #include "testing/gtest/include/gtest/gtest.h" |
| #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h" |
| @@ -151,7 +153,7 @@ int AcmReceiveTestOldApi::RegisterExternalReceiveCodec( |
| } |
| void AcmReceiveTestOldApi::Run() { |
| - for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet; |
| + for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet; |
| packet.reset(packet_source_->NextPacket())) { |
| // Pull audio until time to insert packet. |
| while (clock_.TimeInMilliseconds() < packet->time_ms()) { |